Historically, telecommunications have involved the transmission of signals (e.g. voice) over a network dedicated to telecommunications, such as the public switched telephone network (PSTN) or a private branch exchange (PBX). Similarly, data communications between computers have also historically been transmitted on a dedicated data network, such as a local area network (LAN) or a wide area network (WAN), for example. Generally, telecommunications and data transmissions have been merged into an integrated communication network using technologies such as Voice over Internet Protocol (VoIP).
Audio and/or video streaming across a communication network may encounter delays that diminish the advantages of real-time communications. Jitter is a variable-length delay that can cause a flow (e.g. a conversation) between two or more end points (e.g. two people) to break or to deteriorate and, thus, become unintelligible. Jitter is a variation in the delay of received packets. At the sending side, packets are sent in a continuous stream with the packets being evenly spaced apart. As a result of network congestion, improper queuing, or configuration errors, this steady stream can become fragmented: causing the delay between each packet to vary instead of remaining constant. In VoIP networks in which existing data traffic might be bursty, jitter can be problematic. This could inhibit the successful propagation of any type of real-time (or quasi real-time) communications.
Thus, the ability to properly manage real-time (or quasi real-time) communication flows presents a significant challenge to system designers, component manufacturers, and network operators.